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01992 892323
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IP phone - Splicecom
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| SpliceCom Maximiser is
a business IP phone system. It just happens to be constructed
around a pure IP core because we had the luxury of starting
completely from scratch when we were designing it.
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| The use of this particular
IP phone technology has allowed the delivery of greater
scalability, flexibility and more advanced features
than traditional TDM based, or IP Enabled TDM phone
systems. Of increasing importance to businesses is the
fact that this approach also provides an easy – not
to mention cost effective – way of directly converging
your telephone system with the World Wide Web, intranet
and other web-enabled IT applications. This is the basis
of what the Gartner Group term the Real Time Enterprise;
increasing work force efficiency and reducing end-to-end
business process by using web-based applications to
get the right information to the right people at the
right time. |

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Maximiser uses the PCS 410, PCS 100, PCS Operators Console
and PCS 50 IP Softphone for Windows and Mac OS X to
deliver a sophisticated end to end pure IP phone solution
– where and when required. |
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H.323 and SIP
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No need to choose when you can use
both at the same time with SpliceCom Maximiser, as the
5100 and 5108 Call Servers both support H.323 and the
Session Initiation Protocol (SIP). These two protocols
provide different alternatives for initiating, modifying,
and terminating an interactive user session that involves
multimedia elements – or setting up a Voice over IP
phone call as it’s more commonly known. Support for
SIP, which has its roots in the IP community (it’s similar
to HTTP and shares some of its design principles) from
both manufacturers and service providers is growing.
On the other hand, H.323 is more mature protocol and
has its basis in ISDN. Both H.323 and SIP use the Real
Time Transport Protocol (RTP) to carry the actual media
(voice or video) stream once the call has been set-up.
SpliceCom Maximiser allows both H.323 and SIP devices
to be concurrently supported across a single system,
with calls being passed seamlessly between both types
of terminals. SIP Desktop Phones from Linksys, Polycom,
Grandstream, Elmeg/SNOM & Safecom have been tested with
maximiser, alongside SIP WiFi phones from Hitachi.
H.323 and SIP Trunk services are also supported on SpliceCom
Maximiser through the use of IP Trunk Licences. Although
these Next Generation trunk services are currently in
their infancy, maximiser has already been successfully
tested with SIP services from Gamma Telecom, VoIP Unlimited,
Voiceflex/Frontier Telecom and Gradwell/Margathea Telecom.
Testing has also been successfully completed with Border
Gateway Controllers (BGCs) supplied by Hipcom. maximiser
embraces ISDN, H.323 and SIP trunks as equals, providing
true one-world communications without the need for additional
Gateways, delivering both choice and future-proofing. |
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Migrating to IP Telephony - one step at a time
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| The ideal scenario for the introduction
of IP Telephony is the Greenfield site. As the majority
of companies don't have this “start from scratch” luxury,
maximiser has been developed to integrate with your
current network infrastructure, exactly as it stands,
so maintaining your current investment and enabling
migration to full IP Tphones at your own pace. |
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PBX Replacement
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If you're looking to replace an existing
business telephone system that's run out of steam, then
the investment protection offered to you by the SpliceCom
Maximiser is likely to be a key factor in your decision
making process. You can install the required number
of Maximiser modules, Call Server, Trunk and IP Phones,
in a single cabinet in exactly the same manner as a
traditional PBX.
SpliceCom allows you to retain your existing investment
in copper plant. By terminating your existing twisted
pair cabling with RJ45 connectors they can be directly
connected to the Phone modules. Maximiser allows you
to retain your existing investment in traditional analogue
phones - but provides far greater facilities. A single
connection from the 5100 or 5108 Call Server fed into
your existing LAN is all it takes to further energise
these phones by allowing them to utilise existing desktop
PCs and laptops (Windows, Mac OS X and Linux) to run
the PCS 50 Phone Partner.
The QoS LAN Switch integral to the Call Server ensures
that your voice and data traffic remain on separate
networks. Want to move into the world of IP Telephony?
Then just connect Category 5 cables (or better) directly
to the Maximiser Call Server - or a LAN Switch connected
directly to the Call Server - and your PCS 410s, PCS
100s SIP or H.323/H.450 compliant IP phones. This allows
you to change the telephony mix and migrate to the newer
voice technologies as and when it’s right for your particular
organisational needs, but you can still keep your voice
and data networks separate. |
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Utilising your Existing LAN Infrastructure
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| Do the LAN Switches you're currently
using on your network support Quality of Service (QoS)?
If the answer’s no it doesn't matter - with the SpliceCom
Maximiser you still have full access to ALL the business
benefits it has to offer by keeping your voice and data
traffic separate. However, if you're using a structured
cabling scheme for voice and data connectivity, then
SpliceCom Maximiser can deliver further cost savings
and efficiencies to your organisation. You can now distribute
Call Server, Trunk & Phone modules exactly where they're
needed throughout your building or campus, rather than
being forced to centralise them within a single cabinet
- as is the case with traditional and indeed IP-enabled
PBXs. Now cables for analogue extensions only need to
run from the desk to the riser where the Phone Module
can be located - a single LAN cable handles up to 34
concurrent phone calls between the Phone Module and
the Call Server. A single connection between the Call
Server and your existing LAN network allows the PCS
50 IP Phone Partner to be run on desktop PCs, delivering
all the benefits of the PCS 410 to your analogue phone
users. Again, PCS 410s and PCS 100s can be connected
directly to the Call Server or via low-cost LAN switches
dedicated to voice traffic - this being the case these
switches do not require QoS support. |
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| If your LAN infrastructure equipment
already supports QoS, Maximiser allows you to fully
converge your voice & data networks. Connectivity of
Call Server, Trunk & Phone modules along with PCS 410
and PCS 100 IP phones can be overlaid on top of your
existing IP network. Where the latter two devices are
deployed, only one LAN cable to the desktop is now required
and IP Softphones, such as the PCS 50, running on desktop
or laptop PCs (Windows and Apple Mac OS X) can be utilised.
The upshot of this is only one network for voice & data,
far less equipment to manage and a massive reduction
in the amount of cabling required. |
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Legacy PBX Migration
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Many companies will have made significant investments
in their central site PBX that they're not ready to
write-off just yet. Some will have little desire to
disrupt everyday operations, yet want to trial IP phones
in anticipation of short-to-medium business needs. Others
will have PBXs that cannot be upgraded to support IP
in the future or the cost of an IP phone system upgrade
is deemed excessive. A few may need to bring a new branch
office on-line, but not want to invest further in their
current preferred, yet aging PBX solution. For all of
these scenarios, and many more, Maximiser can be used
in conjunction with your existing telephone system.
Deploying maximiser behind your existing PBX (or fronting
it) can be achieved simply via DPNSS or using PRI or
BRI S Bus connection(s). This allows calls and CLI information
to be automatically routed and manually transferred
between switches. In situations where remote networking
of SpliceCom Maximiser to a central site PBX is required,
this can be achieved over a DPNSS links, or over an
existing IP WAN router infrastructure between offices
with a Call Server acting as a central-site gateway.
In scenarios where no data link between sites already
exists, SpliceCom Maximiser’s integral IP WAN router
can be utilised to enable the transport of inter-site
voice calls and data, between offices, over a single
link.
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